一篇关于MPEG声频编码器与解码器的论文 CAJ文件

关于SDK的 语音识别和语音合成的代码。 可以试试啊-On the SDK Speech Recognition and Speech Synthesis code. Try ah

用matlab仿真实现语音信号处理中分析合成语音的lms算法

自己编写的用于传感器信号滤波的算法程序，并有真实信号处理的数据演示-I have written for the sensor signal filtering algorithms procedures and have the authentic signal processing of data presentation

语音识别中交互式录音的实现，论文，格式为ｖｉｐ-interactive voice recognition recording the realization of papers, the format for vip

基于ADSP TS201S的LMS算法-Based on the LMS algorithm ADSP TS201S

A wonderful ica toollbox about blind source separation for the beginners

Epos, a freely distributable language independent Text-to-Speech system.

共振峰合成非常好用-A very convenient project on formant synthesis.

Levison-Durbin 语音信号处理中的线性预测编码LPC 理论、格型滤波器以及求解现行预 测方程的算法，可以实现对语音信号重要元素的分析、合成甚至识别。 基于现有的实验平台，我们可以利用 Matlab 函数来获得几个固定语音元素（如元音） 的模型系数，LPC 得到的系数组成 IIR 滤波器。利用冲击脉冲 序列作为输入，我们就可以得到原来的语音。这是一种简单的语音合成功能。-Levison-Durbin speech signal processing in linear predictive coding LPC theory, lattice filters, as well as the current prediction equation solving algorithm, can achieve an important element of the speech signal analysis, synthesis or recognition. Based on the existing experimental platform, we can use Matlab function to obtain the number of fixed-voice elements (such as vowels) model coefficients, LPC coefficients are the composition of IIR filters. Shock pulse sequence used as input, we can get the original voice. This is a simple voice synthesis.

时域波形、短时能量、短时平均幅度、短时过零率、短时过电平率-Time-domain waveform, short-term energy, short-term average rate, short-term zero crossing rate, short-term rates over Level